It is known to provide a system for conducting a telephone conference over a circuit-switched telephone network, e.g. a fixed-line network such as a PSTN (public switched telephone network) or POTS (“Plain Old Telephone Service”). “Conference” in this context means a call which can in principle be accessed by three or more participants (users) simultaneously, each accessing the call from a different respective end-point terminal (there are in fact a percentage of conference calls where only two parties dial into a bridge—they're conference calls because they use a conference bridge, even if the access code was given only to two parties, and even if the organizer reserved only two ports on the bridge). To access the conference call, users typically dial in to an audio conference bridge and enter an access code which demonstrates that they are authorised to access the conference, e.g. having been invited by the organiser. Other forms of access control also found, both in addition to and as a substitute for the access code.
It is also known to enable voice or video conference calls over a packet-based communication system such as a VoIP system (“voice over internet protocol”) implemented over an internetwork such as the Internet. Again “conference” means that three or more users at respective end-points can in principle participate simultaneously in the same call. To enable VoIP calls, a communication client application is installed on a terminal (e.g. on a smart phone, tablet, laptop or desktop computer, or even a household media appliance such as a television set or set-top box). (Note: if the terminal is a hardware PBX (VoIP) phone, or a sip-phone, the user installs no software—a hardware VoIP handset typically ships with whatever software it needs so the user installs nothing.) When executed on the terminal, the VoIP client may first enable the user to register a username with a database which maps usernames to IP addresses (either a distributed peer-to-peer database stored amongst multiple end-user terminals and/or a database stored at a server of the VoIP provider). An executed VoIP client may then access this database in order to set-up a call by looking up the IP address for the callee. Each user also might use their client to maintain a respective contact list at a server of the VoIP provider, which lists other users with whom that user has mutually agreed to become contacts such that the user and the contacts are mutually authorised to call one another. The user can select contacts from the contact list to invite into a conference call. Since participation in the conference call is administered by means of the usernames and contact lists, VoIP systems do not conventionally involve the need to dial in an access code.
It is also possible to conduct a call between a VoIP client and a conventional telephone of a circuit-switched network. This is conventionally achieved via a gateway between the circuit-switched network and the packet-switched network. In this case the VoIP user may supply a telephone number from his or her client application to the gateway over the packet-switched network, and the gateway then dials out to the relevant end-point of the circuit-switched network; or the user of the telephone in the circuit-switched network dials a telephone number which switches to the gateway, and the gateway then maps the telephone number to a user of the VoIP system and routes the call to the relevant IP address. If the VoIP user then adds other participants, the call can become a conference call involving both VoIP users and a circuit-switched telephone user.